I'm currently working on generating a tone on a PIC32 device. The information I've found has not been enough to give me a complete understanding of how to achieve this. As I understand it a PWM signal sends 1's and 0's with specified duty cycle and frequency such that it's possible to make something rotate in a certain speed for example. But that to generate a tone this is not enough. I'm primarily focusing on the following two links to create the code:
http://umassamherstm5.org/tech-tutorials/pic32-tutorials/pic32mx220-tutorials/pwm http://www.mikroe.com/chapters/view/54/chapter-6-output-compare-module/#ch6.4
And also the relevant parts in the reference manual.
One of the links states that to play audio it's necessary to use the timer interrupts. How should these be used? Is it necessary to compute the value of the wave with for example a sine function and then combine this with the timer interrupts to define the duty cycle after each interrupt flag?
The end result will be a program that responds to button presses and plays sounds. If a low pass filter is necessary this will be implemented as well.
If you're using PWM to simulate a DAC and output arbitrary audio (for a simple and dirty tone of a given frequency you don't need this complexity), you want to take audio samples (PCM) and convert them each into the respective duty cycle.
Reasonable audio begins at sample rates of 8KHz (POTS). So, for every (every 1/8000th of second) sample you'll need to change the duty cycle. And you want these changes to be regular as irregularities will contribute to audible distortions. So you can program a timer to generate interrupts at 8KHz rate and in the ISR change the duty cycle according to the new audio sample value (this ISR has to read the samples from memory, unless they form a simple pattern and may be computed on the fly).
When you change the duty cycle at a rate of 8KHz you generate a periodic wave at the frequency of 4KHz. This is very well audible. Filtering it well in analogue circuitry without affecting the sound that you want to hear may not be a very easy thing to do (sharp LPF filters are tricky/expensive, cheap filters are poor). Instead you can up the sample rate to either above twice what the speaker can produce (or the human ear can hear) or at least well above the maximum frequency that you want to produce (in this latter case a cheap analogue filter can help rid the unwanted periodic wave without much effect on what you want to hear, you don't need as much sharpness here).
Be warned, if the sample rate is higher than that of your audio file, you'll need a proper upsampler/sample-rate converter. Also remember that raising the sample rate will raise CPU utilization (ISR invoked more times per second, plus sample rate conversion, unless your audio is pre-converted) and power consumption.
[I've done this before on my PC's speaker, but it's now ruined, thanks to SMM/SMIs used by the BIOS and the chipset.]